similar to: How to compile and install just one module?

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

need transcoding to a|ulaw. I am using it with no problems (have g729 licenses as well though). A bit off topic, I have found some extra configuration that is not really in the docs (or I could not find them): fullname=Your full name country=gr language=en city=City province=Province phone_home=+fullinternationalnumber phone_office=+fullinternationalnumber email=your at email.com

2011/6/14 Kevin P. Fleming <kpfleming at digium.com> > On 06/14/2011 03:11 PM, Olivier wrote: > >> Hi, >> >> 1. Is there any manual entry about modprobe's options relative to a >> given Dahdi driver (wctdm24xxp, for instance) ? >> >> 2. When loading a wctdm24xxp driver, is there any parameter to pass to >> modprobe to configure a span in

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you

Kevin, since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about). philippe alex.robar@gmail.com From:

Hello everyone! I installed Asterisk following the instructions of the book "Asterisk: The Future of Telephony". (very nice book) However, I failed. I installed zaptel, libpri and asterisk (in this order). The Installation of Zaptel is successful and my TDM400P is correctly detected: # zttool Alarms Span OK Wildcard S400P

Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

It may have gone to sleep. Chris Cooper Systems/Network Administrator EFC International 1940 Craigshire Blvd St. Louis, MO 63146 US Phone - 314-439-4325 Fax - 314-439-4443 Mobile - 314-402-8912 - -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent:

I've created a page on wiki.asterisk.org outlining some changes we're proposing to make to the Asterisk release and support cycles. As always, before implementing any changes of this type, we'd like to collect some community feedback on the proposal. The page is here: https://wiki.asterisk.org/wiki/x/5ggiAQ Feel free to comment here, or on the page itself if you find any errors

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the

Recently it was brought to our attention that while we announce new releases of Asterisk and Asterisk-Addons on the asterisk-announce mailing list (and others), and we publish security advisories on the asterisk-announce and asterisk-security mailing lists (and others), there are frequently changes that are made in our policies, procedures, and products that do not get announced on any widely-read

In order to more closely align the services that Digium provides to the Asterisk open source community with the Asterisk project itself, we've recently renamed many of the servers that provide these services. Effective immediately: 1) http://bugs.digium.com has moved to https://issues.asterisk.org There are no content or functional changes (except for the new site being SSL/TLS enabled),

All of sudden my FXS module is not working. I have a TDM card with one FXS and one FXO, FXO module seems working fine. I also noticed the LED is not on for my FXS module while it is on for my FXO module. Sep 13 12:11:44 WARNING[9870]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 12:11:44 ERROR[9870]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device